In order to contain costs and extend functionality, enterprises are eager to adopt Voice over IP (VoIP) and other Internet Protocol (IP)-based communication services. To address this need, developers of Private Branch Exchange (PBX) equipment and software have developed IP PBXs to provide the functionality users expect in conventional PBXs, such as conference calling, call forwarding, automatic call distribution, shared message boxes, etc., and to provide new functionality that takes advantage of the multimedia content that can be transmitted over the Internet.
Session Initiation Protocol (SIP), specified in the RFC 3261 of the Internet Engineering Task Force (IETF) SIP Working Group, is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants, and is widely used as a signaling protocol for VoIP. SIP sessions can be of different media types, including Internet Protocol (IP) telephone calls, instant messaging (IM), multimedia distribution, and multimedia conferences. SIP provides a signaling and call setup protocol for IP-based communications that can support many of the call processing functions and features present in the public switched telephone network (PSTN). SIP itself does not define these features. However, SIP permits such features to be built into network elements, such as proxy servers and user agents, and implementing these features permits familiar telephone-like operations, such as dialing a number, causing a phone to ring, and hearing ringback tones or a busy signal.
SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP acts as a carrier for the Session Description Protocol (SDP), which describes the set up and media content of the session, such as the IP ports to use and the codec being used. SIP clients typically use Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) to connect to SIP servers and other SIP endpoints. SIP is most commonly used to set up and tear down voice and video calls. However, it can be used in any application where session management is a requirement, such as event subscription and notification, and terminal mobility. All communications are done over separate session protocols, typically implementing Real-Time Transport Protocol (RTP).
SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7). However, while SS7 is a highly centralized protocol, characterized by complex central network architecture and unintelligent endpoints (conventional telephone handsets), SIP is a peer-to-peer protocol. SIP features are typically implemented in the communicating endpoints (i.e. at the edge of the network) as opposed to traditional SS7 features, which are implemented in the network.
Enterprises, and their employees, increasingly rely on mobile communication devices, such as cellular telephones and wireless messaging devices, to carry out day-to-day business. Currently, WiFi and dual mode mobile devices can access SIP-enabled call processing functions of their related enterprise, but only when the mobile device in question is within range of the related enterprise WiFi network. When the user of a dual mode mobile device roams out of WiFi range of the enterprise network, he is left with cellular communication only. Thus, mobile device users out of their enterprise WiFi range do not have access to the rich IP-enabled features and services, nor do they, typically, have access to the features and services offered by the PBX associated with their enterprise.
It is, therefore, desirable to provide a method and system that provides SIP-enabled call processing functions to mobile devices operating in a cellular network.